Dal post originale:
The Asterisk Development Team is pleased to announce the second beta release of
Asterisk 10.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

With the release of the Asterisk 10 branch, the preceding '1.' has been removed
from the version number per the blog post available at
http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-a...

All interested users of Asterisk are encouraged to participate in the
Asterisk 10 testing process. Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira. It is also very useful to see
successful test reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing
channel on IRC to work together in testing the many parts of Asterisk.

Asterisk 10 is the next major release series of Asterisk. It will be a
Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk
versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

A short list of features includes:

T.38 gateway functionality has been added to res_fax.
Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
Support for defining hints has been added to pbx_lua.
Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
Much, much more!

A full list of new features can be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

For a full list of changes in the current release, please see the ChangeLog:

Dal post originale:
he release of Asterisk 1.8.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

Please note that a significant numbers of changes and fixes have gone into
features.c in this release (call parking, built-in transfers, call pickup,
etc.).

NOTE:

Recently, we were notified that the mechanism included in our Asterisk source
code releases to download and build support for the iLBC codec had stopped
working correctly; a little investigation revealed that this occurred because of
some changes on the ilbcfreeware.org website. These changes occurred as a result
of Google's acquisition of GIPS, who produced (and provided licenses for) the
iLBC codec.

If you are a user of Asterisk and iLBC together, and you've already executed a
license agreement with GIPS, we believe you can continue using iLBC with
Asterisk. If you are a user of Asterisk and iLBC together, but you had not
executed a license agreement with GIPS, we encourage you to research the
situation and consult with your own legal representatives to determine what
actions you may want to take (or avoid taking).

Added the 'storesipcause' option to sip.conf to allow the user to disable the
setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set
HASH(SIP_CAUSE,) on the channel carries a significant performance
penalty because of the usage of the MASTER_CHANNEL() dialplan function.

We've decided to disable this feature by default in future 1.8 versions. This
would be an unexpected behavior change for anyone depending on that SIP_CAUSE
update in their dialplan. Please refer to the asterisk-dev mailing list more
information:

http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
Significant fixes and improvements to parking lots.
(Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452,
ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada,
Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett)
Numerous issues have been reported for deadlocks that are caused by a blocking
read in res_timing_timerfd on a file descriptor that will never be written to.

Dal post originale:
The release of Asterisk 1.8.6.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

Fix an issue with Music on Hold classes losing files in playlist when realtime
is used.
(Closes issue ASTERISK-17875. Reported by David Cunningham. Patched by Igor
Goncharovsky)
Resolve a potential crash in chan_sip when utilizing auth= and performing a
'sip reload' from the console.
(Closes issue ASTERISK-17939. Reported by wdoekes. Patched by Richard Mudgett)
Address some improper sql statements in res_odbc that would cause an update
to fail on realtime peers due to trying to set as "(NULL)" rather than an
actual NULL.
(Closes issue ASTERISK-17791. Reported by marcelloceschia. Patched by Tilghman
Lesher)
Resolve issue where 403 Forbidden would always be sent maximum number of times
regardless to receipt of ACK.
(Patched by Richard Mudgett)
Resolve issue where if a call to MeetMe includes both the dynamic(D) and
always request PIN(P) options, MeetMe will ask for the PIN two times: once for
creating the conference and once for entering the conference.
(Patched by Kinsey Moore)
Fix New Zealand indications profile based on
http://www.telepermit.co.nz/TNA102.pdf
(Closes issue ASTERISK-16263. Reported, Patched by richardf)
Segfault in shell_helper in func_shell.c
(Closes issue ASTERISK-18109. Reported by Michael Myles, patched by Richard
Mudgett)

Dal post originale:
The Asterisk Development Team is pleased to announce the first beta release of
Asterisk 10.0.0-beta1. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

With the release of the Asterisk 10 branch, the preceding '1.' has been removed
from the version number per the blog post available at
http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-a...

All interested users of Asterisk are encouraged to participate in the
Asterisk 10 testing process. Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira. It is also very useful to see
successful test reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing
channel on IRC to work together in testing the many parts of Asterisk.
Additionally users can make use of the RPM and DEB packages now being built for
all Asterisk releases. More information available at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages

Asterisk 10 is the next major release series of Asterisk. It will be a
Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk
versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

A short list of included features includes:

T.38 gateway functionality has been added to res_fax.
Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
Support for defining hints has been added to pbx_lua.
Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
Much, much more!

A full list of new features can be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

For a full list of changes in the current release, please see the ChangeLog:

Dal post originale:
The release of Asterisk 1.8.5.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

Fix Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,
cmaj)
Fixes thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,
rossbeer, kowalma, Freddi_Fonet)
Be more tolerant of what URI we accept for call completion PUBLISH requests.
(Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)
Fix a nasty chanspy bug which was causing a channel leak every time a spied on
channel made a call.
(Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
This patch fixes a bug with MeetMe behavior where the 'P' option for always
prompting for a pin is ignored for the first caller.
(Closes issue #18070. Reported by mav3rick. Patched by bbryant)
Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
the call that the dialplan started an AGI script for is hungup while the AGI
script is in the middle of a command then the AGI script is not notified of
the hangup.
(Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
Resolve issue where leaving a voicemail, the MWI message is never sent. The
same thing happens when checking a voicemail and marking it as read.
(Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard
Mudgett)
Resolve issue where wait for leader with Music On Hold allows crosstalk
between participants. Parenthesis in the wrong position. Regression from issue
#14365 when expanding conference flags to use 64 bits.
(Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)

Dal post originale:
The Asterisk Development Team announces the release of libpri version
1.4.12. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri/

Dal post originale:
The Asterisk Development Team announces the first release candidate of
Asterisk 1.8.5. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.5-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

Fix Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,
cmaj)
Fixes thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,
rossbeer, kowalma, Freddi_Fonet)
Be more tolerant of what URI we accept for call completion PUBLISH requests.
(Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)
Fix a nasty chanspy bug which was causing a channel leak every time a spied on
channel made a call.
(Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
This patch fixes a bug with MeetMe behavior where the 'P' option for always
prompting for a pin is ignored for the first caller.
(Closes issue #18070. Reported by mav3rick. Patched by bbryant)
Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
the call that the dialplan started an AGI script for is hungup while the AGI
script is in the middle of a command then the AGI script is not notified of
the hangup.
(Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
Resolve issue where leaving a voicemail, the MWI message is never sent. The
same thing happens when checking a voicemail and marking it as read.
(Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard
Mudgett)
Resolve issue where wait for leader with Music On Hold allows crosstalk
between participants. Parenthesis in the wrong position. Regression from issue
#14365 when expanding conference flags to use 64 bits.
(Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)
Fix timerfd locking issue.
(Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz)

For a full list of changes in this release candidate, please see the ChangeLog:

Dal post originale:
The Asterisk Development Team has announced the final maintenance release of
Asterisk, version 1.6.2.19. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Please note that Asterisk 1.6.2.19 is the final maintenance release from the
1.6.2 branch. Support for security related issues will continue until April 21,
2012. For more information about support of the various Asterisk branches, see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

The release of Asterisk 1.6.2.19 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

Don't broadcast FullyBooted to every AMI connection
The FullyBooted event should not be sent to every AMI connection
every time someone connects via AMI. It should only be sent to
the user who just connected.
(Closes issue #18168. Reported, patched by FeyFre)
Fix thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Tested by vois, rossbeer, kowalma,
Freddi_Fonet. Patched by dvossel)
Don't delay DTMF in core bridge while listening for DTMF features.
(Closes issue #15642, #16625. Reported by jasonshugart, sharvanek. Tested by
globalnetinc, jde. Patched by oej, twilson)
Fix chan_local crashs in local_fixup()
Thanks OEJ for tracking down the issue and submitting the patch.
(Closes issue #19053. Reported, patched by oej)
Don't offer video to directmedia callee unless caller offered it as well
(Closes issue #19195. Reported, patched by one47)

Additionally security announcements AST-2011-008, AST-2011-010, and
AST-2011-011 have been resolved in this release.

Dal post originale:
he Asterisk Development Team has announced the final maintenance release of
Asterisk, version 1.4.42. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Please note that Asterisk 1.4.42 is the final maintenance release from the
1.4 branch. Support for security related issues will continue until April 21,
2012. For more information about support of the various Asterisk branches, see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

The release of Asterisk 1.4.42 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

Resolve regression with ring groups in the Dial() application
(Closes issue ASTERISK-17874. Reported by mspuhler. Patched by elguero)
Resolve deadlock when using tab completion on the 'meetme kick' CLI command
when an invalid (non-existent) conference room is specified.
(Closes issue ASTERISK-17771. Reported, patched by zvision)
Resolve issue where voice frames could be dropped when checking for T.38
during early media.
(Closes issue ASTERISK-17705. Reported, patched by oej)
Resolve issue where DYNAMIC_FEATURES would not activate after a recent
DTMF fix.
(Closes issue ASTERISK-17914. Reported by vrban. Patched by twilson)

Additionally security announcements AST-2011-010, and AST-2011-011 have been
resolved in this release.

Dal post originale:
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 resolves the
following issue:

AST-2011-011: Asterisk may respond differently to SIP requests from an
invalid SIP user than it does to a user configured on the system, even when the
alwaysauthreject option is set in the configuration. This can leak information
about what SIP users are valid on the Asterisk system.

For more information about the details of this vulnerability, please read
the security advisory AST-2011-011, which was released at the same time as this
announcement.

For a full list of changes in the current releases, please see the ChangeLog: